Historically, entirely separate communication systems have been employed to transmit audio data (sometimes referred to shorthandedly as "voice" for simplicity's sake) and computer data (sometimes abbreviated "data" for a like purpose, although it should be understood that "voice" data and computer "data" both fall within the broad definition of "data" ).
Over a century ago, analog telephone networks were developed to carry analog audio signals. Telephone networks allow communication of audio data, or more broadly, audio signals between two or more users by establishing, with central switching equipment, a dedicated communication circuit or "channel" among the users. Because the channel, once established, is dedicated exclusively to transmission of the users' conversation, the conversation is not required to compete for the channel's bandwidth with other conversations. The advantage of having a dedicated channel per conversation is that any transmission delays from speaker to listener is purely a function of the unfettered speed of the audio signal through the telephone network. Since this speed does not significantly vary over time, such dedicated channels are capable of providing "isochronous" transmission. Unfortunately, one significant disadvantage of dedicated channels is that they require significant bandwidth; that is, the complete bandwidth of the channel remains available and dedicated to carriage of the conversation even when no audio information is being transmitted.
In recent years, efforts have been underway to establish interface standards for digital transmission of audio signals over telephone networks. The most noted of the existing standards is the Integrated Services Digital Network ("ISDN") significantly sponsored by AT&T. ISDN standardizes connection interfaces, transmission protocols and services to create a unified digital circuit switching network. More recently, recommendations for Broadband ISDN ("BISDN") have been adopted. Unlike ISDN, which is a digital network standard, BISDN uses packet relay, or Asynchronous Transfer Mode ("ATM") as a transmission standard, and is of particular importance in transmission over broadband "backbones" and, in particular, fiber optic lines. ATM is primarily a connection-oriented technique that can transport both connection and connectionless-oriented services at either a constant bit rate or a variable bit rate. ATM provides bandwidth on demand and handles all traffic types through fast-packet switching techniques that reduce the processing of protocols and uses statistical multiplexing.
In ATM, audio data are split into relatively small blocks or packets, commonly called "cells." The cells are individually communicated through the ATM network by transmitters and receivers that are not synchronized. Networks limited to synchronous transmission generally require dedicated channels and a clock to control the synchronous transmission of audio data through the network. Therefore, ATM allows telephone networks to depart from the above-described synchronous transmission of audio data over dedicated, isochronous channels, thereby dramatically increasing network efficiency by combining previously dedicated channels and decreasing cost by eliminating synchronicity. Both ISDN and BISDN therefore hold much promise for the future. However, widespread application of these standards has been slow, as the installed base of analog equipment (including telephone sets) is substantial and presents great resistance to change.
Packet transmission or ATM should not be confused with TDM. TDM calls for synchronous division of the overall bandwidth of a common backbone into multiple low speed channels and assigns a specific time slot to each channel. In other words, if there are four channels, each channel is allocated a fourth of the bandwidth. The bandwidth is systematically switched, such that channel 1 gets its fourth-bandwidth, followed by channels 2, 3, 4, 1, 2 and so on. In TDM, the processing power necessary to share common bandwidth is located in various, centralized multiplexers. This centralization is acceptable if channel traffic is constant or predictable. However, when traffic occurs in short intervals (as in the real world), processing becomes nontrivial, resulting in an effective loss of bandwidth.
In contrast, packet transmission or ATM is asynchronous, allocating the total backbone bandwidth on an as-demanded basis. For instance, if channel 1 is highly active, it may receive more than its pro-rata share of overall bandwidth. When channel 1's activity declines, its allocated bandwidth likewise declines. Thus, packet transmission or ATM is most adept at handling "bursty" transmission of data, wherein the activity of each individual channel is subject to relatively wide variation. Thus, because computers transmit data through networks in packets, computer data are said to be "bursty." Unlike TDM, the processing power required to create, transmit and receive packets is distributed among all of the communicating devices, rather than being centralized. Thus, bandwidth is not effectively lost due to inherent limitations in centralized processing.
Although telephone networks have been in place for over a century, computer networks have come into being only in the past quarter century. In contrast with the dedicated channels of traditional telephone networks, computer networks allow individual computers shared access to a common communication backbone having relatively broad bandwidth (in a manner quite similar to ATM).
As in ATM, computer data are divided into packets, each of which includes error protection. The individual networked computers ("nodes") thus are granted access to the complete bandwidth of the backbone so they can transmit their packets of computer data thereon. When the transmitting computer completes transmission of the packet, the backbone is made immediately available for the other computers.
A special case of a computer network is a personal computer ("PC") network. Whereas PCs were once only used as isolated devices, they are now used for a wide range of applications requiring the PCs to communicate with each other over a computer network.
Today, networking in a large office with hundreds of PCs, or in a small office with just a few PCs, is very popular and, quite simply, the best way to share data and communicate among PCs. A local area network ("LAN") is a specific type of network connecting PCs located in relatively close proximity. A wide area network ("WAN") is a network of separate LANs. The backbones of such LANs typically comprise coaxial or twisted-pair cable.
All networks experience delay in end-to-end data transmissions therethrough. This delay (termed "latency") affects the overall efficiency and effective bandwidth of the network. ATM and computer networks, because they are asynchronous, are further subject to "jitter," defined as change in network latency as a function of time. Jitter is largely unpredictable; however, the overall quantity of traffic on a network tends to increase both latency and jitter.
At the heart of any computer network is a communication protocol. A protocol is a set of conventions or rules that govern the transfer of data between computer devices. The simplest protocols define only a hardware configuration, while more complex protocols define timing, data formats, error detection and correction techniques and software structures.
Computer networks almost universally employ multiple layers of protocols. A low-level physical layer protocol assures the transmission and reception of a data stream between two devices. Data packets are constructed in a data link layer. Over the physical layer, a network and transport layer protocol governs transmission of data through the network, thereby ensuring end-to end reliable data delivery.
The most common physical networking protocol or topology for small networks is Ethernet, developed by Xerox. When a node possesses a packet to be transmitted through the network, the node monitors the backbone and transmits when the backbone becomes clear. There is no central backbone master device to grant requests to gain access to the backbone. While this type of multipoint topology facilitates rapid transmission of data when the backbone is lightly utilized, packet collisions may occur when the backbone is heavily utilized. In such circumstances, there is a greater chance that multiple nodes will detect that the backbone is clear and transmit their packets coincidentally. If packets are impaired in a collision, the packets are retransmitted until transmission is successful.
Another conventional physical protocol or topology is Token Ring, developed by IBM. This topology employs a "token" that is passed unidirectionally from node to node around an annular backbone. The node possessing the token is granted exclusive access to the backbone for a single packet transfer. While this topology reduces data collisions, the latency incurred while each node waits for the token translates into a slower data transmission rate than Ethernet when the network is lightly utilized.
Several network and transport protocols designed to handle bursty data transmission are well known in the art. One protocol that enables communication between PCs is the Microcom Networking Protocol ("MNP"), developed by Microcom Systems. MNP is suited for both interactive communication and file transfers and may be implemented on a wide range of computers. MNP packets data with a header and trailer containing packet type, CRC and other information concerning the packet. While the MNP protocol provides relatively error-free transmission of data, the significant overhead of the header and trailer decreases data bandwidth.
The prior art includes many techniques involving manipulation of data to boost the data transmission rate or "throughput" of a network. U.S. Pat. No. 4,691,314, assigned to Microcom, discloses a system for transmitting data in larger, adjustable-sized packets. Because the system allows for larger packets, relatively less header and trailer overhead is required.
However, when the transmission medium is unreliable (such as when the data are transmitted over noisy telephone network lines), errors may occur more frequently in the data. As packet length increases, the chance of corruption of the data within the packet also increases. Furthermore, the larger packets must be retransmitted, thereby decreasing network throughput.
Another network and transport protocol is Transmission Control Protocol/Internet Protocol ("TCP/IP"). This protocol employs a "go back N method" of error and flow control over a datagram network. In a "go back N method" of error control, if there is a transmission error, a packet loss, excessive latency in the delivery of a packet, delivery of a packet out of sequence or an overflow of a receiver buffer, significant loss of throughput is realized due to excessive packet retransmissions.
As the domain of digital computer networks continues to expand, the networks are challenged with new and more difficult responsibilities. One of those challenges is multimedia. In recent years, there have been a number of attempts to produce a digital data network additionally capable of carrying data representing a digitized audio signal (again, "voice"), thereby additionally functioning as a telephone network and, in sum, yielding a so-called "multimedia network."
As described above, however, audio signals are extremely time-sensitive, because users are extremely sensitive to minute tones, inflections and pauses, particularly in human speech. Thus, a computer data network that also must transmit audio data is forced to cope with the communication of both bursty computer and time-sensitive audio data on the backbone.
The repercussion is that the above-described data network and transport protocols that are sufficient to transmit data are insufficient for transmission of time-sensitive audio data. The latencies present in a communication network, e.g., those relating to coding, packet assembly, media access, propagation, receiver buffering and decoding, must be precisely compensated for to preserve the fidelity of the audio signal.
At this point, an interesting observation should be made. Data has been described above as being bursty. It has been implied that audio data is somehow not. Both of these assumptions prove to be inaccurate. First, data is only bursty because computer networks have been dealing with it in that manner for so many years. In fact, once transmission of a batch of data begins, data transmission rate is constant. Second, because spoken words are made of small, discrete utterances (syllables or words), audio data is inherently bursty. Therefore, while it is certainly true that audio data is extremely time-sensitive, audio data is likewise bursty. If a way can be found to compensate for network jitter, audio data should be highly amenable to packet-based transmission.
Therefore, what is needed in the art is a system and method for transmitting and receiving digitized audio data in a packet-based network to adjust for variable packet transmission times. The system and method must deliver end-to-end reliable transmission of data, accounting for all delays in the transmission network while presenting high fidelity audio signals at the receiving end.